Merge tag 'v4.4-rc6' into devel

Linux 4.4-rc6

41 files changed, 520 insertions, 159 deletions

diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c
index 5d99436..0cda05c 100644
--- a/sound/firewire/dice/dice.c
+++ b/sound/firewire/dice/dice.c
@@ -12,9 +12,11 @@ MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_LICENSE("GPL v2");
 
 #define OUI_WEISS		0x001c6a
+#define OUI_LOUD		0x000ff2
 
 #define DICE_CATEGORY_ID	0x04
 #define WEISS_CATEGORY_ID	0x00
+#define LOUD_CATEGORY_ID	0x10
 
 static int dice_interface_check(struct fw_unit *unit)
 {
@@ -57,6 +59,8 @@ static int dice_interface_check(struct fw_unit *unit)
 	}
 	if (vendor == OUI_WEISS)
 		category = WEISS_CATEGORY_ID;
+	else if (vendor == OUI_LOUD)
+		category = LOUD_CATEGORY_ID;
 	else
 		category = DICE_CATEGORY_ID;
 	if (device->config_rom[3] != ((vendor << 8) | category) ||
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 8a7fbdc..bff5c8b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -312,6 +312,10 @@ enum {
 	(AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\
 	 AZX_DCAPS_I915_POWERWELL)
 
+#define AZX_DCAPS_INTEL_BROXTON \
+	(AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\
+	 AZX_DCAPS_I915_POWERWELL)
+
 /* quirks for ATI SB / AMD Hudson */
 #define AZX_DCAPS_PRESET_ATI_SB \
 	(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\
@@ -351,6 +355,8 @@ enum {
 					((pci)->device == 0x0d0c) || \
 					((pci)->device == 0x160c))
 
+#define IS_BROXTON(pci)	((pci)->device == 0x5a98)
+
 static char *driver_short_names[] = {
 	[AZX_DRIVER_ICH] = "HDA Intel",
 	[AZX_DRIVER_PCH] = "HDA Intel PCH",
@@ -502,15 +508,36 @@ static void azx_init_pci(struct azx *chip)
         }
 }
 
+/*
+ * In BXT-P A0, HD-Audio DMA requests is later than expected,
+ * and makes an audio stream sensitive to system latencies when
+ * 24/32 bits are playing.
+ * Adjusting threshold of DMA fifo to force the DMA request
+ * sooner to improve latency tolerance at the expense of power.
+ */
+static void bxt_reduce_dma_latency(struct azx *chip)
+{
+	u32 val;
+
+	val = azx_readl(chip, SKL_EM4L);
+	val &= (0x3 << 20);
+	azx_writel(chip, SKL_EM4L, val);
+}
+
 static void hda_intel_init_chip(struct azx *chip, bool full_reset)
 {
 	struct hdac_bus *bus = azx_bus(chip);
+	struct pci_dev *pci = chip->pci;
 
 	if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
 		snd_hdac_set_codec_wakeup(bus, true);
 	azx_init_chip(chip, full_reset);
 	if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
 		snd_hdac_set_codec_wakeup(bus, false);
+
+	/* reduce dma latency to avoid noise */
+	if (IS_BROXTON(pci))
+		bxt_reduce_dma_latency(chip);
 }
 
 /* calculate runtime delay from LPIB */
@@ -2124,6 +2151,9 @@ static const struct pci_device_id azx_ids[] = {
 	/* Sunrise Point-LP */
 	{ PCI_DEVICE(0x8086, 0x9d70),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE },
+	/* Broxton-P(Apollolake) */
+	{ PCI_DEVICE(0x8086, 0x5a98),
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON },
 	/* Haswell */
 	{ PCI_DEVICE(0x8086, 0x0a0c),
 	  .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL },
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index f8a12ca4..4ef2259 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -778,7 +778,8 @@ static const struct hda_pintbl alienware_pincfgs[] = {
 };
 
 static const struct snd_pci_quirk ca0132_quirks[] = {
-	SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15", QUIRK_ALIENWARE),
+	SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE),
+	SND_PCI_QUIRK(0x1028, 0x0688, "Alienware 17 2015", QUIRK_ALIENWARE),
 	{}
 };
 
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index c8b8ef5..ef19890 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -955,6 +955,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
  */
 
 static const struct hda_device_id snd_hda_id_conexant[] = {
+	HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto),
 	HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto),
 	HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto),
 	HDA_CODEC_ENTRY(0x14f15051, "CX20561 (Hermosa)", patch_conexant_auto),
@@ -972,9 +973,9 @@ static const struct hda_device_id snd_hda_id_conexant[] = {
 	HDA_CODEC_ENTRY(0x14f150ac, "CX20652", patch_conexant_auto),
 	HDA_CODEC_ENTRY(0x14f150b8, "CX20664", patch_conexant_auto),
 	HDA_CODEC_ENTRY(0x14f150b9, "CX20665", patch_conexant_auto),
-	HDA_CODEC_ENTRY(0x14f150f1, "CX20721", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f150f1, "CX21722", patch_conexant_auto),
 	HDA_CODEC_ENTRY(0x14f150f2, "CX20722", patch_conexant_auto),
-	HDA_CODEC_ENTRY(0x14f150f3, "CX20723", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f150f3, "CX21724", patch_conexant_auto),
 	HDA_CODEC_ENTRY(0x14f150f4, "CX20724", patch_conexant_auto),
 	HDA_CODEC_ENTRY(0x14f1510f, "CX20751/2", patch_conexant_auto),
 	HDA_CODEC_ENTRY(0x14f15110, "CX20751/2", patch_conexant_auto),
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 60cd9e7..4b6fb66 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2352,6 +2352,12 @@ static void intel_pin_eld_notify(void *audio_ptr, int port)
 	struct hda_codec *codec = audio_ptr;
 	int pin_nid = port + 0x04;
 
+	/* skip notification during system suspend (but not in runtime PM);
+	 * the state will be updated at resume
+	 */
+	if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0)
+		return;
+
 	check_presence_and_report(codec, pin_nid);
 }
 
@@ -2378,7 +2384,8 @@ static int patch_generic_hdmi(struct hda_codec *codec)
 	 * can cover the codec power request, and so need not set this flag.
 	 * For previous platforms, there is no such power well feature.
 	 */
-	if (is_valleyview_plus(codec) || is_skylake(codec))
+	if (is_valleyview_plus(codec) || is_skylake(codec) ||
+			is_broxton(codec))
 		codec->core.link_power_control = 1;
 
 	if (is_haswell_plus(codec) || is_valleyview_plus(codec)) {
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2f7b065..6c268da 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -111,6 +111,7 @@ struct alc_spec {
 	void (*power_hook)(struct hda_codec *codec);
 #endif
 	void (*shutup)(struct hda_codec *codec);
+	void (*reboot_notify)(struct hda_codec *codec);
 
 	int init_amp;
 	int codec_variant;	/* flag for other variants */
@@ -773,6 +774,25 @@ static inline void alc_shutup(struct hda_codec *codec)
 		snd_hda_shutup_pins(codec);
 }
 
+static void alc_reboot_notify(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (spec && spec->reboot_notify)
+		spec->reboot_notify(codec);
+	else
+		alc_shutup(codec);
+}
+
+/* power down codec to D3 at reboot/shutdown; set as reboot_notify ops */
+static void alc_d3_at_reboot(struct hda_codec *codec)
+{
+	snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3);
+	snd_hda_codec_write(codec, codec->core.afg, 0,
+			    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+	msleep(10);
+}
+
 #define alc_free	snd_hda_gen_free
 
 #ifdef CONFIG_PM
@@ -818,7 +838,7 @@ static const struct hda_codec_ops alc_patch_ops = {
 	.suspend = alc_suspend,
 	.check_power_status = snd_hda_gen_check_power_status,
 #endif
-	.reboot_notify = alc_shutup,
+	.reboot_notify = alc_reboot_notify,
 };
 
 
@@ -1759,6 +1779,7 @@ enum {
 	ALC882_FIXUP_NO_PRIMARY_HP,
 	ALC887_FIXUP_ASUS_BASS,
 	ALC887_FIXUP_BASS_CHMAP,
+	ALC882_FIXUP_DISABLE_AAMIX,
 };
 
 static void alc889_fixup_coef(struct hda_codec *codec,
@@ -1920,6 +1941,8 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec,
 
 static void alc_fixup_bass_chmap(struct hda_codec *codec,
 				 const struct hda_fixup *fix, int action);
+static void alc_fixup_disable_aamix(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action);
 
 static const struct hda_fixup alc882_fixups[] = {
 	[ALC882_FIXUP_ABIT_AW9D_MAX] = {
@@ -2151,6 +2174,10 @@ static const struct hda_fixup alc882_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_bass_chmap,
 	},
+	[ALC882_FIXUP_DISABLE_AAMIX] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_disable_aamix,
+	},
 };
 
 static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -2218,6 +2245,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD),
 	SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1458, 0xa182, "Gigabyte Z170X-UD3", ALC882_FIXUP_DISABLE_AAMIX),
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
 	SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD),
 	SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD),
@@ -4190,6 +4218,8 @@ static void alc_fixup_tpt440_dock(struct hda_codec *codec,
 	struct alc_spec *spec = codec->spec;
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->shutup = alc_no_shutup; /* reduce click noise */
+		spec->reboot_notify = alc_d3_at_reboot; /* reduce noise */
 		spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
 		codec->power_save_node = 0; /* avoid click noises */
 		snd_hda_apply_pincfgs(codec, pincfgs);
@@ -4570,6 +4600,7 @@ enum {
 	ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC,
 	ALC293_FIXUP_DELL1_MIC_NO_PRESENCE,
 	ALC292_FIXUP_TPT440_DOCK,
+	ALC292_FIXUP_TPT440,
 	ALC283_FIXUP_BXBT2807_MIC,
 	ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED,
 	ALC282_FIXUP_ASPIRE_V5_PINS,
@@ -4585,8 +4616,11 @@ enum {
 	ALC288_FIXUP_DISABLE_AAMIX,
 	ALC292_FIXUP_DELL_E7X,
 	ALC292_FIXUP_DISABLE_AAMIX,
+	ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK,
 	ALC298_FIXUP_DELL1_MIC_NO_PRESENCE,
 	ALC275_FIXUP_DELL_XPS,
+	ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE,
+	ALC293_FIXUP_LENOVO_SPK_NOISE,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -5041,6 +5075,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST
 	},
+	[ALC292_FIXUP_TPT440] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_disable_aamix,
+		.chained = true,
+		.chain_id = ALC292_FIXUP_TPT440_DOCK,
+	},
 	[ALC283_FIXUP_BXBT2807_MIC] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -5140,6 +5180,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE
 	},
+	[ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_disable_aamix,
+		.chained = true,
+		.chain_id = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE
+	},
 	[ALC292_FIXUP_DELL_E7X] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_dell_xps13,
@@ -5167,6 +5213,23 @@ static const struct hda_fixup alc269_fixups[] = {
 			{}
 		}
 	},
+	[ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			/* Disable pass-through path for FRONT 14h */
+			{0x20, AC_VERB_SET_COEF_INDEX, 0x36},
+			{0x20, AC_VERB_SET_PROC_COEF, 0x1737},
+			{}
+		},
+		.chained = true,
+		.chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
+	},
+	[ALC293_FIXUP_LENOVO_SPK_NOISE] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_disable_aamix,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_THINKPAD_ACPI
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -5180,8 +5243,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
 	SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
 	SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
+	SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
 	SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),
+	SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X),
 	SND_PCI_QUIRK(0x1028, 0x05ca, "Dell Latitude E7240", ALC292_FIXUP_DELL_E7X),
 	SND_PCI_QUIRK(0x1028, 0x05cb, "Dell Latitude E7440", ALC292_FIXUP_DELL_E7X),
 	SND_PCI_QUIRK(0x1028, 0x05da, "Dell Vostro 5460", ALC290_FIXUP_SUBWOOFER),
@@ -5199,11 +5264,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
-	SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
-	SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
-	SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
-	SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
-	SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
+	SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
+	SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
+	SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
+	SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
+	SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
+	SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE),
 	SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -5302,15 +5368,17 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
-	SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440),
 	SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2211, "Thinkpad W541", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE),
 	SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP),
 	SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -5320,6 +5388,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE),
 	SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
@@ -5400,6 +5469,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"},
 	{.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"},
 	{.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"},
+	{.id = ALC292_FIXUP_TPT440, .name = "tpt440"},
 	{}
 };
 
@@ -6386,6 +6456,7 @@ static const struct hda_fixup alc662_fixups[] = {
 static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2),
 	SND_PCI_QUIRK(0x1025, 0x022f, "Acer Aspire One", ALC662_FIXUP_INV_DMIC),
+	SND_PCI_QUIRK(0x1025, 0x0241, "Packard Bell DOTS", ALC662_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 826122d..2c7c5eb 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -3110,6 +3110,29 @@ static void stac92hd71bxx_fixup_hp_hdx(struct hda_codec *codec,
 	spec->gpio_led = 0x08;
 }
 
+static bool is_hp_output(struct hda_codec *codec, hda_nid_t pin)
+{
+	unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, pin);
+
+	/* count line-out, too, as BIOS sets often so */
+	return get_defcfg_connect(pin_cfg) != AC_JACK_PORT_NONE &&
+		(get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT ||
+		 get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT);
+}
+
+static void fixup_hp_headphone(struct hda_codec *codec, hda_nid_t pin)
+{
+	unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, pin);
+
+	/* It was changed in the BIOS to just satisfy MS DTM.
+	 * Lets turn it back into slaved HP
+	 */
+	pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE)) |
+		(AC_JACK_HP_OUT << AC_DEFCFG_DEVICE_SHIFT);
+	pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC | AC_DEFCFG_SEQUENCE))) |
+		0x1f;
+	snd_hda_codec_set_pincfg(codec, pin, pin_cfg);
+}
 
 static void stac92hd71bxx_fixup_hp(struct hda_codec *codec,
 				   const struct hda_fixup *fix, int action)
@@ -3119,22 +3142,12 @@ static void stac92hd71bxx_fixup_hp(struct hda_codec *codec,
 	if (action != HDA_FIXUP_ACT_PRE_PROBE)
 		return;
 
-	if (hp_blike_system(codec->core.subsystem_id)) {
-		unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f);
-		if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT ||
-			get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER  ||
-			get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT) {
-			/* It was changed in the BIOS to just satisfy MS DTM.
-			 * Lets turn it back into slaved HP
-			 */
-			pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE))
-					| (AC_JACK_HP_OUT <<
-						AC_DEFCFG_DEVICE_SHIFT);
-			pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC
-							| AC_DEFCFG_SEQUENCE)))
-								| 0x1f;
-			snd_hda_codec_set_pincfg(codec, 0x0f, pin_cfg);
-		}
+	/* when both output A and F are assigned, these are supposedly
+	 * dock and built-in headphones; fix both pin configs
+	 */
+	if (is_hp_output(codec, 0x0a) && is_hp_output(codec, 0x0f)) {
+		fixup_hp_headphone(codec, 0x0a);
+		fixup_hp_headphone(codec, 0x0f);
 	}
 
 	if (find_mute_led_cfg(codec, 1))
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index 714df90..41c31db 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -741,10 +741,11 @@ snd_rme96_playback_setrate(struct rme96 *rme96,
 	{
 		/* change to/from double-speed: reset the DAC (if available) */
 		snd_rme96_reset_dac(rme96);
+		return 1; /* need to restore volume */
 	} else {
 		writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+		return 0;
 	}
-	return 0;
 }
 
 static int
@@ -980,6 +981,7 @@ snd_rme96_playback_hw_params(struct snd_pcm_substream *substream,
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	int err, rate, dummy;
+	bool apply_dac_volume = false;
 
 	runtime->dma_area = (void __force *)(rme96->iobase +
 					     RME96_IO_PLAY_BUFFER);
@@ -993,24 +995,26 @@ snd_rme96_playback_hw_params(struct snd_pcm_substream *substream,
 	{
                 /* slave clock */
                 if ((int)params_rate(params) != rate) {
-			spin_unlock_irq(&rme96->lock);
-			return -EIO;                    
-                }
-	} else if ((err = snd_rme96_playback_setrate(rme96, params_rate(params))) < 0) {
-		spin_unlock_irq(&rme96->lock);
-		return err;
-	}
-	if ((err = snd_rme96_playback_setformat(rme96, params_format(params))) < 0) {
-		spin_unlock_irq(&rme96->lock);
-		return err;
+			err = -EIO;
+			goto error;
+		}
+	} else {
+		err = snd_rme96_playback_setrate(rme96, params_rate(params));
+		if (err < 0)
+			goto error;
+		apply_dac_volume = err > 0; /* need to restore volume later? */
 	}
+
+	err = snd_rme96_playback_setformat(rme96, params_format(params));
+	if (err < 0)
+		goto error;
 	snd_rme96_setframelog(rme96, params_channels(params), 1);
 	if (rme96->capture_periodsize != 0) {
 		if (params_period_size(params) << rme96->playback_frlog !=
 		    rme96->capture_periodsize)
 		{
-			spin_unlock_irq(&rme96->lock);
-			return -EBUSY;
+			err = -EBUSY;
+			goto error;
 		}
 	}
 	rme96->playback_periodsize =
@@ -1021,9 +1025,16 @@ snd_rme96_playback_hw_params(struct snd_pcm_substream *substream,
 		rme96->wcreg &= ~(RME96_WCR_PRO | RME96_WCR_DOLBY | RME96_WCR_EMP);
 		writel(rme96->wcreg |= rme96->wcreg_spdif_stream, rme96->iobase + RME96_IO_CONTROL_REGISTER);
 	}
+
+	err = 0;
+ error:
 	spin_unlock_irq(&rme96->lock);
-		
-	return 0;
+	if (apply_dac_volume) {
+		usleep_range(3000, 10000);
+		snd_rme96_apply_dac_volume(rme96);
+	}
+
+	return err;
 }
 
 static int
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 9929efc..b3ea24d 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1023,24 +1023,18 @@ void arizona_init_dvfs(struct arizona_priv *priv)
 }
 EXPORT_SYMBOL_GPL(arizona_init_dvfs);
 
-static unsigned int arizona_sysclk_48k_rates[] = {
+static unsigned int arizona_opclk_ref_48k_rates[] = {
 	6144000,
 	12288000,
 	24576000,
 	49152000,
-	73728000,
-	98304000,
-	147456000,
 };
 
-static unsigned int arizona_sysclk_44k1_rates[] = {
+static unsigned int arizona_opclk_ref_44k1_rates[] = {
 	5644800,
 	11289600,
 	22579200,
 	45158400,
-	67737600,
-	90316800,
-	135475200,
 };
 
 static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk,
@@ -1065,11 +1059,11 @@ static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk,
 	}
 
 	if (refclk % 8000)
-		rates = arizona_sysclk_44k1_rates;
+		rates = arizona_opclk_ref_44k1_rates;
 	else
-		rates = arizona_sysclk_48k_rates;
+		rates = arizona_opclk_ref_48k_rates;
 
-	for (ref = 0; ref < ARRAY_SIZE(arizona_sysclk_48k_rates) &&
+	for (ref = 0; ref < ARRAY_SIZE(arizona_opclk_ref_48k_rates) &&
 		     rates[ref] <= refclk; ref++) {
 		div = 1;
 		while (rates[ref] / div >= freq && div < 32) {
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 969e337..84f5eb0 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -205,18 +205,18 @@ static const struct snd_kcontrol_new es8328_right_line_controls =
 
 /* Left Mixer */
 static const struct snd_kcontrol_new es8328_left_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 8, 1, 0),
-	SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 7, 1, 0),
-	SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 8, 1, 0),
-	SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 7, 1, 0),
+	SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 7, 1, 0),
+	SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 6, 1, 0),
+	SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 7, 1, 0),
+	SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 6, 1, 0),
 };
 
 /* Right Mixer */
 static const struct snd_kcontrol_new es8328_right_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 8, 1, 0),
-	SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 7, 1, 0),
-	SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 8, 1, 0),
-	SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 7, 1, 0),
+	SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 7, 1, 0),
+	SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 6, 1, 0),
+	SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 7, 1, 0),
+	SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 6, 1, 0),
 };
 
 static const char * const es8328_pga_sel[] = {
diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c
index 7fc7b4e..c1b87c5 100644
--- a/sound/soc/codecs/nau8825.c
+++ b/sound/soc/codecs/nau8825.c
@@ -1271,6 +1271,36 @@ static int nau8825_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
+#ifdef CONFIG_PM_SLEEP
+static int nau8825_suspend(struct device *dev)
+{
+	struct i2c_client *client = to_i2c_client(dev);
+	struct nau8825 *nau8825 = dev_get_drvdata(dev);
+
+	disable_irq(client->irq);
+	regcache_cache_only(nau8825->regmap, true);
+	regcache_mark_dirty(nau8825->regmap);
+
+	return 0;
+}
+
+static int nau8825_resume(struct device *dev)
+{
+	struct i2c_client *client = to_i2c_client(dev);
+	struct nau8825 *nau8825 = dev_get_drvdata(dev);
+
+	regcache_cache_only(nau8825->regmap, false);
+	regcache_sync(nau8825->regmap);
+	enable_irq(client->irq);
+
+	return 0;
+}
+#endif
+
+static const struct dev_pm_ops nau8825_pm = {
+	SET_SYSTEM_SLEEP_PM_OPS(nau8825_suspend, nau8825_resume)
+};
+
 static const struct i2c_device_id nau8825_i2c_ids[] = {
 	{ "nau8825", 0 },
 	{ }
@@ -1297,6 +1327,7 @@ static struct i2c_driver nau8825_driver = {
 		.name = "nau8825",
 		.of_match_table = of_match_ptr(nau8825_of_ids),
 		.acpi_match_table = ACPI_PTR(nau8825_acpi_match),
+		.pm = &nau8825_pm,
 	},
 	.probe = nau8825_i2c_probe,
 	.remove = nau8825_i2c_remove,
diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c
index aca479f..1dc68ab 100644
--- a/sound/soc/codecs/rl6231.c
+++ b/sound/soc/codecs/rl6231.c
@@ -80,8 +80,10 @@ int rl6231_calc_dmic_clk(int rate)
 	}
 
 	for (i = 0; i < ARRAY_SIZE(div); i++) {
-		/* find divider that gives DMIC frequency below 3MHz */
-		if (3000000 * div[i] >= rate)
+		if ((div[i] % 3) == 0)
+			continue;
+		/* find divider that gives DMIC frequency below 3.072MHz */
+		if (3072000 * div[i] >= rate)
 			return i;
 	}
 
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 2813237..ef76940 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -245,7 +245,7 @@ struct rt5645_priv {
 	struct snd_soc_jack *hp_jack;
 	struct snd_soc_jack *mic_jack;
 	struct snd_soc_jack *btn_jack;
-	struct delayed_work jack_detect_work;
+	struct delayed_work jack_detect_work, rcclock_work;
 	struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)];
 	struct rt5645_eq_param_s *eq_param;
 
@@ -565,12 +565,33 @@ static int rt5645_hweq_put(struct snd_kcontrol *kcontrol,
 	.put = rt5645_hweq_put \
 }
 
+static int rt5645_spk_put_volsw(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+	struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component);
+	int ret;
+
+	cancel_delayed_work_sync(&rt5645->rcclock_work);
+
+	regmap_update_bits(rt5645->regmap, RT5645_MICBIAS,
+		RT5645_PWR_CLK25M_MASK, RT5645_PWR_CLK25M_PU);
+
+	ret = snd_soc_put_volsw(kcontrol, ucontrol);
+
+	queue_delayed_work(system_power_efficient_wq, &rt5645->rcclock_work,
+		msecs_to_jiffies(200));
+
+	return ret;
+}
+
 static const struct snd_kcontrol_new rt5645_snd_controls[] = {
 	/* Speaker Output Volume */
 	SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL,
 		RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1),
-	SOC_DOUBLE_TLV("Speaker Playback Volume", RT5645_SPK_VOL,
-		RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv),
+	SOC_DOUBLE_EXT_TLV("Speaker Playback Volume", RT5645_SPK_VOL,
+		RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, snd_soc_get_volsw,
+		rt5645_spk_put_volsw, out_vol_tlv),
 
 	/* ClassD modulator Speaker Gain Ratio */
 	SOC_SINGLE_TLV("Speaker ClassD Playback Volume", RT5645_SPO_CLSD_RATIO,
@@ -1498,7 +1519,7 @@ static void hp_amp_power(struct snd_soc_codec *codec, int on)
 				regmap_write(rt5645->regmap, RT5645_PR_BASE +
 					RT5645_MAMP_INT_REG2, 0xfc00);
 				snd_soc_write(codec, RT5645_DEPOP_M2, 0x1140);
-				msleep(40);
+				msleep(70);
 				rt5645->hp_on = true;
 			} else {
 				/* depop parameters */
@@ -3122,6 +3143,15 @@ static void rt5645_jack_detect_work(struct work_struct *work)
 				SND_JACK_BTN_2 | SND_JACK_BTN_3);
 }
 
+static void rt5645_rcclock_work(struct work_struct *work)
+{
+	struct rt5645_priv *rt5645 =
+		container_of(work, struct rt5645_priv, rcclock_work.work);
+
+	regmap_update_bits(rt5645->regmap, RT5645_MICBIAS,
+		RT5645_PWR_CLK25M_MASK, RT5645_PWR_CLK25M_PD);
+}
+
 static irqreturn_t rt5645_irq(int irq, void *data)
 {
 	struct rt5645_priv *rt5645 = data;
@@ -3348,6 +3378,27 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "Reks"),
 		},
 	},
+	{
+		.ident = "Google Edgar",
+		.callback = strago_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Edgar"),
+		},
+	},
+	{
+		.ident = "Google Wizpig",
+		.callback = strago_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Wizpig"),
+		},
+	},
+	{
+		.ident = "Google Terra",
+		.callback = strago_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Terra"),
+		},
+	},
 	{ }
 };
 
@@ -3587,6 +3638,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
 	}
 
 	INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work);
+	INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work);
 
 	if (rt5645->i2c->irq) {
 		ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq,
@@ -3621,6 +3673,7 @@ static int rt5645_i2c_remove(struct i2c_client *i2c)
 		free_irq(i2c->irq, rt5645);
 
 	cancel_delayed_work_sync(&rt5645->jack_detect_work);
+	cancel_delayed_work_sync(&rt5645->rcclock_work);
 
 	snd_soc_unregister_codec(&i2c->dev);
 	regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies);
diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h
index dc2b462..3f1b0f1 100644
--- a/sound/soc/codecs/rt5670.h
+++ b/sound/soc/codecs/rt5670.h
@@ -973,12 +973,12 @@
 #define RT5670_SCLK_SRC_MCLK			(0x0 << 14)
 #define RT5670_SCLK_SRC_PLL1			(0x1 << 14)
 #define RT5670_SCLK_SRC_RCCLK			(0x2 << 14) /* 15MHz */
-#define RT5670_PLL1_SRC_MASK			(0x3 << 12)
-#define RT5670_PLL1_SRC_SFT			12
-#define RT5670_PLL1_SRC_MCLK			(0x0 << 12)
-#define RT5670_PLL1_SRC_BCLK1			(0x1 << 12)
-#define RT5670_PLL1_SRC_BCLK2			(0x2 << 12)
-#define RT5670_PLL1_SRC_BCLK3			(0x3 << 12)
+#define RT5670_PLL1_SRC_MASK			(0x7 << 11)
+#define RT5670_PLL1_SRC_SFT			11
+#define RT5670_PLL1_SRC_MCLK			(0x0 << 11)
+#define RT5670_PLL1_SRC_BCLK1			(0x1 << 11)
+#define RT5670_PLL1_SRC_BCLK2			(0x2 << 11)
+#define RT5670_PLL1_SRC_BCLK3			(0x3 << 11)
 #define RT5670_PLL1_PD_MASK			(0x1 << 3)
 #define RT5670_PLL1_PD_SFT			3
 #define RT5670_PLL1_PD_1			(0x0 << 3)
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 1f590b5..2828591 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -1386,90 +1386,90 @@ static const struct snd_kcontrol_new rt5677_dac_r_mix[] = {
 };
 
 static const struct snd_kcontrol_new rt5677_sto1_dac_l_mix[] = {
-	SOC_DAPM_SINGLE("ST L Switch", RT5677_STO1_DAC_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("ST L Switch", RT5677_STO1_DAC_MIXER,
 			RT5677_M_ST_DAC1_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER,
 			RT5677_M_DAC1_L_STO_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_STO1_DAC_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC2 L Switch", RT5677_STO1_DAC_MIXER,
 			RT5677_M_DAC2_L_STO_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER,
 			RT5677_M_DAC1_R_STO_L_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5677_sto1_dac_r_mix[] = {
-	SOC_DAPM_SINGLE("ST R Switch", RT5677_STO1_DAC_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("ST R Switch", RT5677_STO1_DAC_MIXER,
 			RT5677_M_ST_DAC1_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER,
 			RT5677_M_DAC1_R_STO_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_STO1_DAC_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC2 R Switch", RT5677_STO1_DAC_MIXER,
 			RT5677_M_DAC2_R_STO_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER,
 			RT5677_M_DAC1_L_STO_R_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5677_mono_dac_l_mix[] = {
-	SOC_DAPM_SINGLE("ST L Switch", RT5677_MONO_DAC_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("ST L Switch", RT5677_MONO_DAC_MIXER,
 			RT5677_M_ST_DAC2_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_MONO_DAC_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC1 L Switch", RT5677_MONO_DAC_MIXER,
 			RT5677_M_DAC1_L_MONO_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER,
 			RT5677_M_DAC2_L_MONO_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER,
 			RT5677_M_DAC2_R_MONO_L_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5677_mono_dac_r_mix[] = {
-	SOC_DAPM_SINGLE("ST R Switch", RT5677_MONO_DAC_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("ST R Switch", RT5677_MONO_DAC_MIXER,
 			RT5677_M_ST_DAC2_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_MONO_DAC_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC1 R Switch", RT5677_MONO_DAC_MIXER,
 			RT5677_M_DAC1_R_MONO_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER,
 			RT5677_M_DAC2_R_MONO_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER,
 			RT5677_M_DAC2_L_MONO_R_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5677_dd1_l_mix[] = {
-	SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5677_DD1_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix L Switch", RT5677_DD1_MIXER,
 			RT5677_M_STO_L_DD1_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("Mono DAC Mix L Switch", RT5677_DD1_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix L Switch", RT5677_DD1_MIXER,
 			RT5677_M_MONO_L_DD1_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC3 L Switch", RT5677_DD1_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC3 L Switch", RT5677_DD1_MIXER,
 			RT5677_M_DAC3_L_DD1_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC3 R Switch", RT5677_DD1_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC3 R Switch", RT5677_DD1_MIXER,
 			RT5677_M_DAC3_R_DD1_L_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5677_dd1_r_mix[] = {
-	SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5677_DD1_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix R Switch", RT5677_DD1_MIXER,
 			RT5677_M_STO_R_DD1_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("Mono DAC Mix R Switch", RT5677_DD1_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix R Switch", RT5677_DD1_MIXER,
 			RT5677_M_MONO_R_DD1_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC3 R Switch", RT5677_DD1_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC3 R Switch", RT5677_DD1_MIXER,
 			RT5677_M_DAC3_R_DD1_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC3 L Switch", RT5677_DD1_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC3 L Switch", RT5677_DD1_MIXER,
 			RT5677_M_DAC3_L_DD1_R_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5677_dd2_l_mix[] = {
-	SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5677_DD2_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix L Switch", RT5677_DD2_MIXER,
 			RT5677_M_STO_L_DD2_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("Mono DAC Mix L Switch", RT5677_DD2_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix L Switch", RT5677_DD2_MIXER,
 			RT5677_M_MONO_L_DD2_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC4 L Switch", RT5677_DD2_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC4 L Switch", RT5677_DD2_MIXER,
 			RT5677_M_DAC4_L_DD2_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC4 R Switch", RT5677_DD2_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC4 R Switch", RT5677_DD2_MIXER,
 			RT5677_M_DAC4_R_DD2_L_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5677_dd2_r_mix[] = {
-	SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5677_DD2_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix R Switch", RT5677_DD2_MIXER,
 			RT5677_M_STO_R_DD2_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("Mono DAC Mix R Switch", RT5677_DD2_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix R Switch", RT5677_DD2_MIXER,
 			RT5677_M_MONO_R_DD2_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC4 R Switch", RT5677_DD2_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC4 R Switch", RT5677_DD2_MIXER,
 			RT5677_M_DAC4_R_DD2_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC4 L Switch", RT5677_DD2_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC4 L Switch", RT5677_DD2_MIXER,
 			RT5677_M_DAC4_L_DD2_R_SFT, 1, 1),
 };
 
@@ -2596,6 +2596,21 @@ static int rt5677_vref_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int rt5677_filter_power_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		msleep(50);
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
 static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT,
 		0, rt5677_set_pll1_event, SND_SOC_DAPM_PRE_PMU |
@@ -3072,19 +3087,26 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
 
 	/* DAC Mixer */
 	SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5677_PWR_DIG2,
-		RT5677_PWR_DAC_S1F_BIT, 0, NULL, 0),
+		RT5677_PWR_DAC_S1F_BIT, 0, rt5677_filter_power_event,
+		SND_SOC_DAPM_POST_PMU),
 	SND_SOC_DAPM_SUPPLY("dac mono2 left filter", RT5677_PWR_DIG2,
-		RT5677_PWR_DAC_M2F_L_BIT, 0, NULL, 0),
+		RT5677_PWR_DAC_M2F_L_BIT, 0, rt5677_filter_power_event,
+		SND_SOC_DAPM_POST_PMU),
 	SND_SOC_DAPM_SUPPLY("dac mono2 right filter", RT5677_PWR_DIG2,
-		RT5677_PWR_DAC_M2F_R_BIT, 0, NULL, 0),
+		RT5677_PWR_DAC_M2F_R_BIT, 0, rt5677_filter_power_event,
+		SND_SOC_DAPM_POST_PMU),
 	SND_SOC_DAPM_SUPPLY("dac mono3 left filter", RT5677_PWR_DIG2,
-		RT5677_PWR_DAC_M3F_L_BIT, 0, NULL, 0),
+		RT5677_PWR_DAC_M3F_L_BIT, 0, rt5677_filter_power_event,
+		SND_SOC_DAPM_POST_PMU),
 	SND_SOC_DAPM_SUPPLY("dac mono3 right filter", RT5677_PWR_DIG2,
-		RT5677_PWR_DAC_M3F_R_BIT, 0, NULL, 0),
+		RT5677_PWR_DAC_M3F_R_BIT, 0, rt5677_filter_power_event,
+		SND_SOC_DAPM_POST_PMU),
 	SND_SOC_DAPM_SUPPLY("dac mono4 left filter", RT5677_PWR_DIG2,
-		RT5677_PWR_DAC_M4F_L_BIT, 0, NULL, 0),
+		RT5677_PWR_DAC_M4F_L_BIT, 0, rt5677_filter_power_event,
+		SND_SOC_DAPM_POST_PMU),
 	SND_SOC_DAPM_SUPPLY("dac mono4 right filter", RT5677_PWR_DIG2,
-		RT5677_PWR_DAC_M4F_R_BIT, 0, NULL, 0),
+		RT5677_PWR_DAC_M4F_R_BIT, 0, rt5677_filter_power_event,
+		SND_SOC_DAPM_POST_PMU),
 
 	SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0,
 		rt5677_sto1_dac_l_mix, ARRAY_SIZE(rt5677_sto1_dac_l_mix)),
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 0563753..5380798 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -229,7 +229,7 @@ SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
 SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
 	6, 1, 0),
 SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
-	7, 1, 0),
+	7, 1, 1),
 
 SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
 	       WM8960_INBMIX1, 4, 7, 0, lineinboost_tlv),
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index b563d67..6a49b36 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -365,8 +365,8 @@ static const struct reg_default wm8962_reg[] = {
 	{ 16924, 0x0059 },   /* R16924 - HDBASS_PG_1 */
 	{ 16925, 0x999A },   /* R16925 - HDBASS_PG_0 */
 
-	{ 17048, 0x0083 },   /* R17408 - HPF_C_1 */
-	{ 17049, 0x98AD },   /* R17409 - HPF_C_0 */
+	{ 17408, 0x0083 },   /* R17408 - HPF_C_1 */
+	{ 17409, 0x98AD },   /* R17409 - HPF_C_0 */
 
 	{ 17920, 0x007F },   /* R17920 - ADCL_RETUNE_C1_1 */
 	{ 17921, 0xFFFF },   /* R17921 - ADCL_RETUNE_C1_0 */
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 4495a40..c1c9c2e 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -681,8 +681,8 @@ static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai,
 	}
 
 	mcasp->tdm_slots = slots;
-	mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = rx_mask;
-	mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = tx_mask;
+	mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask;
+	mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = rx_mask;
 	mcasp->slot_width = slot_width;
 
 	return davinci_mcasp_set_ch_constraints(mcasp);
@@ -908,6 +908,14 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream,
 		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
 		mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
 			       FSRMOD(total_slots), FSRMOD(0x1FF));
+		/*
+		 * If McASP is set to be TX/RX synchronous and the playback is
+		 * not running already we need to configure the TX slots in
+		 * order to have correct FSX on the bus
+		 */
+		if (mcasp_is_synchronous(mcasp) && !mcasp->channels)
+			mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
+				       FSXMOD(total_slots), FSXMOD(0x1FF));
 	}
 
 	return 0;
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 19c302b..14dfdee 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -283,6 +283,8 @@ config SND_SOC_IMX_MC13783
 config SND_SOC_FSL_ASOC_CARD
 	tristate "Generic ASoC Sound Card with ASRC support"
 	depends on OF && I2C
+	# enforce SND_SOC_FSL_ASOC_CARD=m if SND_AC97_CODEC=m:
+	depends on SND_AC97_CODEC || SND_AC97_CODEC=n
 	select SND_SOC_IMX_AUDMUX
 	select SND_SOC_IMX_PCM_DMA
 	select SND_SOC_FSL_ESAI
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index a4435f5..ffd5f9a 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -454,7 +454,8 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
 	 * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx.
 	 * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx.
 	 */
-	regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0);
+	regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC,
+		           sai->synchronous[TX] ? FSL_SAI_CR2_SYNC : 0);
 	regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC,
 			   sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0);
 
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 7b778ab..d430ef5 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -144,7 +144,7 @@ config SND_SOC_INTEL_SKYLAKE
 
 config SND_SOC_INTEL_SKL_RT286_MACH
 	tristate "ASoC Audio driver for SKL with RT286 I2S mode"
-	depends on X86 && ACPI
+	depends on X86 && ACPI && I2C
 	select SND_SOC_INTEL_SST
 	select SND_SOC_INTEL_SKYLAKE
 	select SND_SOC_RT286
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index a7854c8..ffea427 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -1240,6 +1240,7 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus)
 	 */
 	ret = snd_soc_tplg_component_load(&platform->component,
 					&skl_tplg_ops, fw, 0);
+	release_firmware(fw);
 	if (ret < 0) {
 		dev_err(bus->dev, "tplg component load failed%d\n", ret);
 		return -EINVAL;
diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c
index a38a302..ac72ff5 100644
--- a/sound/soc/rockchip/rockchip_spdif.c
+++ b/sound/soc/rockchip/rockchip_spdif.c
@@ -280,7 +280,7 @@ static int rk_spdif_probe(struct platform_device *pdev)
 	int ret;
 
 	match = of_match_node(rk_spdif_match, np);
-	if ((int) match->data == RK_SPDIF_RK3288) {
+	if (match->data == (void *)RK_SPDIF_RK3288) {
 		struct regmap *grf;
 
 		grf = syscon_regmap_lookup_by_phandle(np, "rockchip,grf");
diff --git a/sound/soc/rockchip/rockchip_spdif.h b/sound/soc/rockchip/rockchip_spdif.h
index 07f86a2..921b409 100644
--- a/sound/soc/rockchip/rockchip_spdif.h
+++ b/sound/soc/rockchip/rockchip_spdif.h
@@ -28,9 +28,9 @@
 #define SPDIF_CFGR_VDW(x)	(x << SPDIF_CFGR_VDW_SHIFT)
 #define SDPIF_CFGR_VDW_MASK	(0xf << SPDIF_CFGR_VDW_SHIFT)
 
-#define SPDIF_CFGR_VDW_16	SPDIF_CFGR_VDW(0x00)
-#define SPDIF_CFGR_VDW_20	SPDIF_CFGR_VDW(0x01)
-#define SPDIF_CFGR_VDW_24	SPDIF_CFGR_VDW(0x10)
+#define SPDIF_CFGR_VDW_16	SPDIF_CFGR_VDW(0x0)
+#define SPDIF_CFGR_VDW_20	SPDIF_CFGR_VDW(0x1)
+#define SPDIF_CFGR_VDW_24	SPDIF_CFGR_VDW(0x2)
 
 /*
  * DMACR
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 76da762..edcf4cc 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -235,7 +235,7 @@ static int rsnd_gen2_probe(struct platform_device *pdev,
 		RSND_GEN_S_REG(SCU_SYS_STATUS0,	0x1c8),
 		RSND_GEN_S_REG(SCU_SYS_INT_EN0,	0x1cc),
 		RSND_GEN_S_REG(SCU_SYS_STATUS1,	0x1d0),
-		RSND_GEN_S_REG(SCU_SYS_INT_EN1,	0x1c4),
+		RSND_GEN_S_REG(SCU_SYS_INT_EN1,	0x1d4),
 		RSND_GEN_M_REG(SRC_SWRSR,	0x200,	0x40),
 		RSND_GEN_M_REG(SRC_SRCIR,	0x204,	0x40),
 		RSND_GEN_M_REG(SRC_ADINR,	0x214,	0x40),
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 261b502..68b439e 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -923,6 +923,7 @@ static int rsnd_src_pcm_new_gen2(struct rsnd_mod *mod,
 			    struct snd_soc_pcm_runtime *rtd)
 {
 	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
+	struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io);
 	struct rsnd_src *src = rsnd_mod_to_src(mod);
 	int ret;
 
@@ -937,6 +938,12 @@ static int rsnd_src_pcm_new_gen2(struct rsnd_mod *mod,
 		return 0;
 
 	/*
+	 * SRC In doesn't work if DVC was enabled
+	 */
+	if (dvc && !rsnd_io_is_play(io))
+		return 0;
+
+	/*
 	 * enable sync convert
 	 */
 	ret = rsnd_kctrl_new_s(mod, io, rtd,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 24b0960..a1305f8 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -795,12 +795,12 @@ static void soc_resume_deferred(struct work_struct *work)
 
 	dev_dbg(card->dev, "ASoC: resume work completed\n");
 
-	/* userspace can access us now we are back as we were before */
-	snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0);
-
 	/* Recheck all endpoints too, their state is affected by suspend */
 	dapm_mark_endpoints_dirty(card);
 	snd_soc_dapm_sync(&card->dapm);
+
+	/* userspace can access us now we are back as we were before */
+	snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0);
 }
 
 /* powers up audio subsystem after a suspend */
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 016eba1..7d00942 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2293,6 +2293,12 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w)
 	kfree(w);
 }
 
+void snd_soc_dapm_reset_cache(struct snd_soc_dapm_context *dapm)
+{
+	dapm->path_sink_cache.widget = NULL;
+	dapm->path_source_cache.widget = NULL;
+}
+
 /* free all dapm widgets and resources */
 static void dapm_free_widgets(struct snd_soc_dapm_context *dapm)
 {
@@ -2303,6 +2309,7 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm)
 			continue;
 		snd_soc_dapm_free_widget(w);
 	}
+	snd_soc_dapm_reset_cache(dapm);
 }
 
 static struct snd_soc_dapm_widget *dapm_find_widget(
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index ecd38e5..2f67ba6 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -404,7 +404,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx);
 /**
  * snd_soc_put_volsw_sx - double mixer set callback
  * @kcontrol: mixer control
- * @uinfo: control element information
+ * @ucontrol: control element information
  *
  * Callback to set the value of a double mixer control that spans 2 registers.
  *
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 8d7ec80..6963ba2 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -531,7 +531,7 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr,
 		/* TLV bytes controls need standard kcontrol info handler,
 		 * TLV callback and extended put/get handlers.
 		 */
-		k->info = snd_soc_bytes_info;
+		k->info = snd_soc_bytes_info_ext;
 		k->tlv.c = snd_soc_bytes_tlv_callback;
 
 		ext_ops = tplg->bytes_ext_ops;
@@ -1805,6 +1805,7 @@ void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm,
 		snd_soc_tplg_widget_remove(w);
 		snd_soc_dapm_free_widget(w);
 	}
+	snd_soc_dapm_reset_cache(dapm);
 }
 EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_remove_all);
 
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index 843f037..5c2bc53 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -669,6 +669,7 @@ static int uni_player_startup(struct snd_pcm_substream *substream,
 {
 	struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai);
 	struct uniperif *player = priv->dai_data.uni;
+	player->substream = substream;
 
 	player->clk_adj = 0;
 
@@ -950,6 +951,8 @@ static void uni_player_shutdown(struct snd_pcm_substream *substream,
 	if (player->state != UNIPERIF_STATE_STOPPED)
 		/* Stop the player */
 		uni_player_stop(player);
+
+	player->substream = NULL;
 }
 
 static int uni_player_parse_dt_clk_glue(struct platform_device *pdev,
@@ -989,7 +992,7 @@ static int uni_player_parse_dt(struct platform_device *pdev,
 	if (!info)
 		return -ENOMEM;
 
-	if (of_property_read_u32(pnode, "version", &player->ver) ||
+	if (of_property_read_u32(pnode, "st,version", &player->ver) ||
 	    player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
 		dev_err(dev, "Unknown uniperipheral version ");
 		return -EINVAL;
@@ -998,13 +1001,13 @@ static int uni_player_parse_dt(struct platform_device *pdev,
 	if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
 		info->underflow_enabled = 1;
 
-	if (of_property_read_u32(pnode, "uniperiph-id", &info->id)) {
+	if (of_property_read_u32(pnode, "st,uniperiph-id", &info->id)) {
 		dev_err(dev, "uniperipheral id not defined");
 		return -EINVAL;
 	}
 
 	/* Read the device mode property */
-	if (of_property_read_string(pnode, "mode", &mode)) {
+	if (of_property_read_string(pnode, "st,mode", &mode)) {
 		dev_err(dev, "uniperipheral mode not defined");
 		return -EINVAL;
 	}
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index f791239..8a0eb20 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -316,7 +316,7 @@ static int uni_reader_parse_dt(struct platform_device *pdev,
 	if (!info)
 		return -ENOMEM;
 
-	if (of_property_read_u32(node, "version", &reader->ver) ||
+	if (of_property_read_u32(node, "st,version", &reader->ver) ||
 	    reader->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
 		dev_err(&pdev->dev, "Unknown uniperipheral version ");
 		return -EINVAL;
@@ -346,7 +346,6 @@ int uni_reader_init(struct platform_device *pdev,
 	reader->hw = &uni_reader_pcm_hw;
 	reader->dai_ops = &uni_reader_dai_ops;
 
-	dev_err(reader->dev, "%s: enter\n", __func__);
 	ret = uni_reader_parse_dt(pdev, reader);
 	if (ret < 0) {
 		dev_err(reader->dev, "Failed to parse DeviceTree");
diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
index bcbf4da..1bb896d 100644
--- a/sound/soc/sunxi/sun4i-codec.c
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -2,6 +2,7 @@
  * Copyright 2014 Emilio López <emilio@elopez.com.ar>
  * Copyright 2014 Jon Smirl <jonsmirl@gmail.com>
  * Copyright 2015 Maxime Ripard <maxime.ripard@free-electrons.com>
+ * Copyright 2015 Adam Sampson <ats@offog.org>
  *
  * Based on the Allwinner SDK driver, released under the GPL.
  *
@@ -404,7 +405,7 @@ static const struct snd_kcontrol_new sun4i_codec_pa_mute =
 static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1);
 
 static const struct snd_kcontrol_new sun4i_codec_widgets[] = {
-	SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL,
+	SOC_SINGLE_TLV("Power Amplifier Volume", SUN4I_CODEC_DAC_ACTL,
 		       SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0,
 		       sun4i_codec_pa_volume_scale),
 };
@@ -452,12 +453,12 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("Mixer Enable", SUN4I_CODEC_DAC_ACTL,
 			    SUN4I_CODEC_DAC_ACTL_MIXEN, 0, NULL, 0),
 
-	/* Pre-Amplifier */
-	SND_SOC_DAPM_MIXER("Pre-Amplifier", SUN4I_CODEC_ADC_ACTL,
+	/* Power Amplifier */
+	SND_SOC_DAPM_MIXER("Power Amplifier", SUN4I_CODEC_ADC_ACTL,
 			   SUN4I_CODEC_ADC_ACTL_PA_EN, 0,
 			   sun4i_codec_pa_mixer_controls,
 			   ARRAY_SIZE(sun4i_codec_pa_mixer_controls)),
-	SND_SOC_DAPM_SWITCH("Pre-Amplifier Mute", SND_SOC_NOPM, 0, 0,
+	SND_SOC_DAPM_SWITCH("Power Amplifier Mute", SND_SOC_NOPM, 0, 0,
 			    &sun4i_codec_pa_mute),
 
 	SND_SOC_DAPM_OUTPUT("HP Right"),
@@ -480,16 +481,16 @@ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = {
 	{ "Left Mixer", NULL, "Mixer Enable" },
 	{ "Left Mixer", "Left DAC Playback Switch", "Left DAC" },
 
-	/* Pre-Amplifier Mixer Routes */
-	{ "Pre-Amplifier", "Mixer Playback Switch", "Left Mixer" },
-	{ "Pre-Amplifier", "Mixer Playback Switch", "Right Mixer" },
-	{ "Pre-Amplifier", "DAC Playback Switch", "Left DAC" },
-	{ "Pre-Amplifier", "DAC Playback Switch", "Right DAC" },
+	/* Power Amplifier Routes */
+	{ "Power Amplifier", "Mixer Playback Switch", "Left Mixer" },
+	{ "Power Amplifier", "Mixer Playback Switch", "Right Mixer" },
+	{ "Power Amplifier", "DAC Playback Switch", "Left DAC" },
+	{ "Power Amplifier", "DAC Playback Switch", "Right DAC" },
 
-	/* PA -> HP path */
-	{ "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" },
-	{ "HP Right", NULL, "Pre-Amplifier Mute" },
-	{ "HP Left", NULL, "Pre-Amplifier Mute" },
+	/* Headphone Output Routes */
+	{ "Power Amplifier Mute", "Switch", "Power Amplifier" },
+	{ "HP Right", NULL, "Power Amplifier Mute" },
+	{ "HP Left", NULL, "Power Amplifier Mute" },
 };
 
 static struct snd_soc_codec_driver sun4i_codec_codec = {
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 7661616..5b4c58c 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -174,6 +174,8 @@ struct snd_usb_midi_in_endpoint {
 		u8 running_status_length;
 	} ports[0x10];
 	u8 seen_f5;
+	bool in_sysex;
+	u8 last_cin;
 	u8 error_resubmit;
 	int current_port;
 };
@@ -468,6 +470,39 @@ static void snd_usbmidi_maudio_broken_running_status_input(
 }
 
 /*
+ * QinHeng CH345 is buggy: every second packet inside a SysEx has not CIN 4
+ * but the previously seen CIN, but still with three data bytes.
+ */
+static void ch345_broken_sysex_input(struct snd_usb_midi_in_endpoint *ep,
+				     uint8_t *buffer, int buffer_length)
+{
+	unsigned int i, cin, length;
+
+	for (i = 0; i + 3 < buffer_length; i += 4) {
+		if (buffer[i] == 0 && i > 0)
+			break;
+		cin = buffer[i] & 0x0f;
+		if (ep->in_sysex &&
+		    cin == ep->last_cin &&
+		    (buffer[i + 1 + (cin == 0x6)] & 0x80) == 0)
+			cin = 0x4;
+#if 0
+		if (buffer[i + 1] == 0x90) {
+			/*
+			 * Either a corrupted running status or a real note-on
+			 * message; impossible to detect reliably.
+			 */
+		}
+#endif
+		length = snd_usbmidi_cin_length[cin];
+		snd_usbmidi_input_data(ep, 0, &buffer[i + 1], length);
+		ep->in_sysex = cin == 0x4;
+		if (!ep->in_sysex)
+			ep->last_cin = cin;
+	}
+}
+
+/*
  * CME protocol: like the standard protocol, but SysEx commands are sent as a
  * single USB packet preceded by a 0x0F byte.
  */
@@ -660,6 +695,12 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = {
 	.output_packet = snd_usbmidi_output_standard_packet,
 };
 
+static struct usb_protocol_ops snd_usbmidi_ch345_broken_sysex_ops = {
+	.input = ch345_broken_sysex_input,
+	.output = snd_usbmidi_standard_output,
+	.output_packet = snd_usbmidi_output_standard_packet,
+};
+
 /*
  * AKAI MPD16 protocol:
  *
@@ -1341,6 +1382,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi *umidi,
 		 * Various chips declare a packet size larger than 4 bytes, but
 		 * do not actually work with larger packets:
 		 */
+	case USB_ID(0x0a67, 0x5011): /* Medeli DD305 */
 	case USB_ID(0x0a92, 0x1020): /* ESI M4U */
 	case USB_ID(0x1430, 0x474b): /* RedOctane GH MIDI INTERFACE */
 	case USB_ID(0x15ca, 0x0101): /* Textech USB Midi Cable */
@@ -2378,6 +2420,10 @@ int snd_usbmidi_create(struct snd_card *card,
 
 		err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
 		break;
+	case QUIRK_MIDI_CH345:
+		umidi->usb_protocol_ops = &snd_usbmidi_ch345_broken_sysex_ops;
+		err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
+		break;
 	default:
 		dev_err(&umidi->dev->dev, "invalid quirk type %d\n",
 			quirk->type);
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index f494dce..4f85757 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1354,6 +1354,8 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 		}
 	}
 
+	snd_usb_mixer_fu_apply_quirk(state->mixer, cval, unitid, kctl);
+
 	range = (cval->max - cval->min) / cval->res;
 	/*
 	 * Are there devices with volume range more than 255? I use a bit more
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index 6a803ef..ddca654 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -348,13 +348,6 @@ static struct usbmix_name_map bose_companion5_map[] = {
 	{ 0 }	/* terminator */
 };
 
-/* Dragonfly DAC 1.2, the dB conversion factor is 1 instead of 256 */
-static struct usbmix_dB_map dragonfly_1_2_dB = {0, 5000};
-static struct usbmix_name_map dragonfly_1_2_map[] = {
-	{ 7, NULL, .dB = &dragonfly_1_2_dB },
-	{ 0 }	/* terminator */
-};
-
 /*
  * Control map entries
  */
@@ -470,11 +463,6 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
 		.id = USB_ID(0x05a7, 0x1020),
 		.map = bose_companion5_map,
 	},
-	{
-		/* Dragonfly DAC 1.2 */
-		.id = USB_ID(0x21b4, 0x0081),
-		.map = dragonfly_1_2_map,
-	},
 	{ 0 } /* terminator */
 };
 
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index fe91184..0ce888d 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -37,6 +37,7 @@
 #include <sound/control.h>
 #include <sound/hwdep.h>
 #include <sound/info.h>
+#include <sound/tlv.h>
 
 #include "usbaudio.h"
 #include "mixer.h"
@@ -1825,3 +1826,39 @@ void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer,
 	}
 }
 
+static void snd_dragonfly_quirk_db_scale(struct usb_mixer_interface *mixer,
+					 struct snd_kcontrol *kctl)
+{
+	/* Approximation using 10 ranges based on output measurement on hw v1.2.
+	 * This seems close to the cubic mapping e.g. alsamixer uses. */
+	static const DECLARE_TLV_DB_RANGE(scale,
+		 0,  1, TLV_DB_MINMAX_ITEM(-5300, -4970),
+		 2,  5, TLV_DB_MINMAX_ITEM(-4710, -4160),
+		 6,  7, TLV_DB_MINMAX_ITEM(-3884, -3710),
+		 8, 14, TLV_DB_MINMAX_ITEM(-3443, -2560),
+		15, 16, TLV_DB_MINMAX_ITEM(-2475, -2324),
+		17, 19, TLV_DB_MINMAX_ITEM(-2228, -2031),
+		20, 26, TLV_DB_MINMAX_ITEM(-1910, -1393),
+		27, 31, TLV_DB_MINMAX_ITEM(-1322, -1032),
+		32, 40, TLV_DB_MINMAX_ITEM(-968, -490),
+		41, 50, TLV_DB_MINMAX_ITEM(-441, 0),
+	);
+
+	usb_audio_info(mixer->chip, "applying DragonFly dB scale quirk\n");
+	kctl->tlv.p = scale;
+	kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+	kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK;
+}
+
+void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer,
+				  struct usb_mixer_elem_info *cval, int unitid,
+				  struct snd_kcontrol *kctl)
+{
+	switch (mixer->chip->usb_id) {
+	case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */
+		if (unitid == 7 && cval->min == 0 && cval->max == 50)
+			snd_dragonfly_quirk_db_scale(mixer, kctl);
+		break;
+	}
+}
+
diff --git a/sound/usb/mixer_quirks.h b/sound/usb/mixer_quirks.h
index bdbfab0..177c329 100644
--- a/sound/usb/mixer_quirks.h
+++ b/sound/usb/mixer_quirks.h
@@ -9,5 +9,9 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip,
 void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer,
 				    int unitid);
 
+void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer,
+				  struct usb_mixer_elem_info *cval, int unitid,
+				  struct snd_kcontrol *kctl);
+
 #endif /* SND_USB_MIXER_QUIRKS_H */
 
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 1a1e2e4..c60a776 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2829,6 +2829,17 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	.idProduct = 0x1020,
 },
 
+/* QinHeng devices */
+{
+	USB_DEVICE(0x1a86, 0x752d),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.vendor_name = "QinHeng",
+		.product_name = "CH345",
+		.ifnum = 1,
+		.type = QUIRK_MIDI_CH345
+	}
+},
+
 /* KeithMcMillen Stringport */
 {
 	USB_DEVICE(0x1f38, 0x0001),
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 5ca80e7..b6c0c8e 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -538,6 +538,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
 		[QUIRK_MIDI_CME] = create_any_midi_quirk,
 		[QUIRK_MIDI_AKAI] = create_any_midi_quirk,
 		[QUIRK_MIDI_FTDI] = create_any_midi_quirk,
+		[QUIRK_MIDI_CH345] = create_any_midi_quirk,
 		[QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk,
 		[QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk,
 		[QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk,
@@ -1124,6 +1125,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
 	case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */
 	case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */
 	case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */
+	case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */
 		return true;
 	}
 	return false;
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 15a1271..b665d85 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -95,6 +95,7 @@ enum quirk_type {
 	QUIRK_MIDI_AKAI,
 	QUIRK_MIDI_US122L,
 	QUIRK_MIDI_FTDI,
+	QUIRK_MIDI_CH345,
 	QUIRK_AUDIO_STANDARD_INTERFACE,
 	QUIRK_AUDIO_FIXED_ENDPOINT,
 	QUIRK_AUDIO_EDIROL_UAXX,